[asterisk-users] Error Configuring Asterisk (FREEPBX)
Hi all, I've just installed again my Asterisk using Xorcom repositories. I can make extensions, but when using any extension i want to dial anything, I got 404 not found using Xlite. Any ideas of what can be happening? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error Configuring Asterisk (FREEPBX)
2007/7/18, Jared Smith [EMAIL PROTECTED]: On Wed, 2007-07-18 at 12:07 -0500, Diego Quintana Cruz wrote: Hi all, I've just installed again my Asterisk using Xorcom repositories. I can make extensions, but when using any extension i want to dial anything, I got 404 not found using Xlite. My guess is that your user (or friend) account in sip.conf is not pointing to the dialplan context that contains the extension you're trying to dial. For example, your sip.conf setting probably looks something like: [xlite] type=friend host=dynamic secret=yoursecretpassword context=blah and then in extensions.conf, you'd have a context that looks like: [blah] exten = 123,1,Playback(hello-world) Whenever a call comes into Asterisk, it first comes through a channel driver (SIP, in this case) and that channel configuration then points the call toward a specific dialplan context. In my examples above, the configuration in sip.conf points to the context named blah in the dialplan. Thanks for your response Jared, that's exactly what is happening, but my asterisk doesn't load my proper dialplan defined in extensions.conf. show dialplan from-internal returns: voip*CLI show dialplan from-internal [ Context 'from-internal' created by 'pbx_config' ] Include ='handle-it' [pbx_config] Include ='phones' [pbx_config] Include ='phone' [pbx_config] Include ='trunk-9' [pbx_config] Include ='features'[pbx_config] Include ='operator'[pbx_config] -= 0 extensions (0 priorities) in 1 context. =- None of those subcontexts are especified, however in my extensions.conf i got [from-internal] include = parkedcalls include = from-internal-custom include = ext-fax include = from-internal-additional include = ext-local-confirm exten = s,1,Macro(hangupcall) exten = h,1,Macro(hangupcall) any ideas? -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error Configuring Asterisk (FREEPBX)
2007/7/18, Diego Quintana Cruz [EMAIL PROTECTED]: 2007/7/18, Jared Smith [EMAIL PROTECTED]: On Wed, 2007-07-18 at 12:07 -0500, Diego Quintana Cruz wrote: Hi all, I've just installed again my Asterisk using Xorcom repositories. I can make extensions, but when using any extension i want to dial anything, I got 404 not found using Xlite. My guess is that your user (or friend) account in sip.conf is not pointing to the dialplan context that contains the extension you're trying to dial. For example, your sip.conf setting probably looks something like: [xlite] type=friend host=dynamic secret=yoursecretpassword context=blah and then in extensions.conf, you'd have a context that looks like: [blah] exten = 123,1,Playback(hello-world) Whenever a call comes into Asterisk, it first comes through a channel driver (SIP, in this case) and that channel configuration then points the call toward a specific dialplan context. In my examples above, the configuration in sip.conf points to the context named blah in the dialplan. Thanks for your response Jared, that's exactly what is happening, but my asterisk doesn't load my proper dialplan defined in extensions.conf. show dialplan from-internal returns: voip*CLI show dialplan from-internal [ Context 'from-internal' created by 'pbx_config' ] Include ='handle-it' [pbx_config] Include ='phones' [pbx_config] Include ='phone' [pbx_config] Include ='trunk-9' [pbx_config] Include ='features'[pbx_config] Include ='operator'[pbx_config] -= 0 extensions (0 priorities) in 1 context. =- None of those subcontexts are especified, however in my extensions.conf i got [from-internal] include = parkedcalls include = from-internal-custom include = ext-fax include = from-internal-additional include = ext-local-confirm exten = s,1,Macro(hangupcall) exten = h,1,Macro(hangupcall) any ideas? Hi all, it was a problem with asterisk-config package, because it kept some files in extensions.d/ directory. One of those files was from-internal.conf. I erased all those files and worked perfectly Regards! -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk mysql support
Hi all, I've just realized that my asterisk isn't storing cdr inputs in mysql. cdr_mysql.conf is well configured and I don't know what else should i configure. I'm using Xorcom's packages, cdr status shows: voip*CLI cdr status CDR logging: enabled CDR mode: simple CDR registered backend: csv CDR registered backend: cdr_manager CDR registered backend: cdr-custom it doesn't appear cdr_mysql. Any ideas? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SugarCRM Integration
Hi folks, I was wondering if there's a guide on how to configure sugarCRM Integration with Asterisk. I was looking in google and all i found was about Trixbox, which has sugarcrm integrated by default. Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 sniffer and player
Hi all, I was wondering if there is any IAX2 sniffer and decoder. Wireshark can decode and play RTP streams using G.711, and Cain Abel decodes and plays any kind of RTP stream. But I didn't find anyone that can decode IAX2 streams. Any programs?? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Areski CDR
Hi folks, I was wondering what happened to Areski CDR viewer that came before with Freepbx. I've noticed that the live-CD contains Areski but the repositories don't have it. Will you fix that? or shall I install Areski from sources? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Nagios/Cacti Plugin
Is this for asterisk 1.2 or asterisk 1.4? 2007/4/26, bkruse [EMAIL PROTECTED]: Hey guys, In my spare time(off of work, not digium related whatsoever) I finished the cacti php script. I need someone to help me do some finishing touches and make a basic layout and pretty colors for the template. All the grunt work and data sources are there, just need to put them into graphs and make them look nice and what not. If your interested in helping/doing this for me, email me at: [EMAIL PROTECTED] Thanks Guys! so far this plugin is Rockin! -bkruse ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ZAPTEL PROBLEM
Hi all, I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything seems nice, but i'm not able to make calls nor to receive any. When I try to make a call, I keep receiven the all circuits are busy now message, and when I receive calls, asterisk doesn't seems to care (don't get anything on the CLI) I'm using Asterisk 1.2.17 and Zaptel 1.2.16 from Xorcom's repository voip:~# asterisk -rx 'zap show status' Description Alarms IRQbpviol CRC4 Wildcard TDM400P REV I Board 1 OK 0 0 0 voip:~# asterisk -rx 'zap show channels' Chan Extension Context Language MusicOnHold pseudofrom-internal es 1from-internal es 2from-internal es 3from-pstn es 4from-pstn es I thought it could be an IRQ problem, but everything seems fine voip:~# cat /proc/interrupts CPU0 0: 118621819 XT-PIC timer 1:811 XT-PIC i8042 2: 0 XT-PIC cascade 5: 0 XT-PIC uhci_hcd:usb3, via82cxxx 6: 5 XT-PIC floppy 7: 0 XT-PIC parport0 8: 1 XT-PIC rtc 9: 0 XT-PIC acpi 10:2879759 XT-PIC uhci_hcd:usb2, eth0 11:3048189 XT-PIC uhci_hcd:usb1, eth1 12: 474378440 XT-PIC ehci_hcd:usb4, wctdm 14:1074418 XT-PIC ide0 15:4239765 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 Hope you can help me with my problem. -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAPTEL PROBLEM
2007/4/30, Tzafrir Cohen [EMAIL PROTECTED]: On Mon, Apr 30, 2007 at 12:25:07PM -0500, Diego Quintana Cruz wrote: Hi all, I am using a TDM400P card with 2 FXS and 2 FXO modules. Everything seems nice, but i'm not able to make calls nor to receive any. When I try to make a call, I keep receiven the all circuits are busy now message, and when I receive calls, asterisk doesn't seems to care (don't get anything on the CLI) set verbose 3? Call from where? To where? From PSTN to Asterisk and viceversa Do you see the relevant channel as offhook in 'zap show channel N' ? I'm not able to to see the channel anymore. voip*CLI zap show channel 3 Unable to find given channel 3 I found that this error happens every time i receive an inbound call: Apr 30 15:08:39 NOTICE[6003] chan_zap.c: Got ZT_EVENT_REMOVED. Destroying channel 3 Sanity check: asterisk -rx 'show channels' voip*CLI show channels Channel Location State Application(Data) 0 active channels 0 active calls voip*CLI zap show channels Chan Extension Context Language MusicOnHold pseudofrom-internal es 1from-internal es 2from-internal es 4from-zaptel es (hmm... asterisk -n -rx 'show channels'hangs for you as well?) I'm using Asterisk 1.2.17 and Zaptel 1.2.16 from Xorcom's repository asterisk-classic or asterisk-bristuff? asterisk-classic voip:~# asterisk -rx 'zap show status' Description Alarms IRQbpviol CRC4 Wildcard TDM400P REV I Board 1 OK 0 0 0 voip:~# asterisk -rx 'zap show channels' Chan Extension Context Language MusicOnHold pseudofrom-internal es 1from-internal es 2from-internal es 3from-pstn es 4from-pstn es I thought it could be an IRQ problem, but everything seems fine voip:~# cat /proc/interrupts CPU0 0: 118621819 XT-PIC timer 1:811 XT-PIC i8042 2: 0 XT-PIC cascade 5: 0 XT-PIC uhci_hcd:usb3, via82cxxx 6: 5 XT-PIC floppy 7: 0 XT-PIC parport0 8: 1 XT-PIC rtc 9: 0 XT-PIC acpi 10:2879759 XT-PIC uhci_hcd:usb2, eth0 11:3048189 XT-PIC uhci_hcd:usb1, eth1 12: 474378440 XT-PIC ehci_hcd:usb4, wctdm ehci_hcd:usb4 does normally take all the USB interrupts. However this issue is probably not related to missed interrupts , if there are any. 14:1074418 XT-PIC ide0 15:4239765 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 Any help would be appreciated -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk - Cisco Call Manager Express Trunk
2007/4/19, Noah Miller [EMAIL PROTECTED]: Hi Diego - I want to make a SIP trunk between a Cisco 2811 router and a Asterisk. Both 2811 and Asterisk are working fine (2811 has 1XX and asterisk 2XX). Now I want to configure a trunk so that 2811 users can call * users. I've been reading a lot but I'm still confused. I don't know if you've seen this page on the WIKI yet, but it does have a section for Call Manger Express: http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration - Noah Hi Noah, Thanks a lot for the page!! -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk - Cisco Call Manager Express Trunk
Hi all, I want to make a SIP trunk between a Cisco 2811 router and a Asterisk. Both 2811 and Asterisk are working fine (2811 has 1XX and asterisk 2XX). Now I want to configure a trunk so that 2811 users can call * users. I've been reading a lot but I'm still confused. Hope you can help me, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FreePBX - Vicidial Integration
Hi all, I am trying to install Vicidial in an existent FreePBX installation (I'm using Xorcom packages for Debian Etch), but I didn't find any documentation, I found only this guide [0], but is for trixbox only, do you think it will work on FreePBX on Etch? [0] http://iptn.org/vicidial/index.html Regards, Diego Quintana Cruz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk HA
2007/1/11, Ale [EMAIL PROTECTED]: Ciao, Enrico Pasqualotto wrote: Is better ultramonkey, dundi or SER proxy in front of * server? You can also consider Hartbeat + rsync, or simply pfsync + rsync ;) The problem with Asterisk HA, is mainly the lost of calls when failover occurs. This is because all traffic pass through Asterisk always. In order to solve this, you could use SER + Asterisk + OpenSER. That way, you'll only lose calls that are going outside your network, but calls inside will remain. -- Diego Quintana a.k.a. RouterMaN Ingeniería de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://planeta.debianperu.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error uninstalling freepbx-panel
Hi everybody, I've installed future packages (asterisk 1.2 and freepbx) from Xorcom's Repository in a debian etch, but when i want to uninstall freepbx-panel, i got this error: dialer:~# apt-get remove --purge freepbx-panel Leyendo lista de paquetes... Hecho Creando árbol de dependencias... Hecho Los siguientes paquetes se ELIMINARÃN: freepbx-panel* 0 actualizados, 0 se instalarán, 1 para eliminar y 1 no actualizados. Necesito descargar 0B de archivos. Se liberarán 65.5kB despuÃ(c)s de desempaquetar. ¿Desea continuar [S/n]? (Leyendo la base de datos ... 105470 ficheros y directorios instalados actualmente.) Desinstalando freepbx-panel ... invoke-rc.d: syntax error: missing required parameter dpkg: error al procesar freepbx-panel (--purge): el subproceso pre-removal script devolvió el código de salida de error 103 Se encontraron errores al procesar: freepbx-panel E: Sub-process /usr/bin/dpkg returned an error code (1) Any ideas how to fix this? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniería de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://planeta.debianperu.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPP problem
2006/9/2, Greg Boehnlein [EMAIL PROTECTED]: On Sat, 2 Sep 2006, Diego Quintana Cruz wrote: Hi everybody, I'm trying to load-test my Asterisk PBX using SIPP, but I always getting errors, I followed the instructions given in [1] which mainly was to create the user sipp in sip.conf and the dialing plan for his context in extensions.conf I'm using Asterisk 1.0.10 Any ideas or tutorial on how using SIP? Here are my notes on the subject: http://lists.digium.com/pipermail/asterisk-dev/2006-June/021162.html I did what you have there but I'm always getting 503 Service unavailable, I don't know why. I'm also using AMPortal, do I have to configure something there? Regards, and sorry for my bad english -- Diego Quintana a.k.a. RouterMaN Ingeniería de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://planeta.debianperu.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIPP problem
For the record, we use a very similar test environment for Asterisk on the Blackfin: * Astersik 1.0.11 (latest Rapid stable debs) - or 1.2.9/1.2.10 from our unstable debs * Diego does most of the job ;-) Anyway, I suggest that you re-read that page. You basically need to alightly eit the supplied sip.conf to match your settings, and also play a bit with sipp (package sip-tester on Debian). Yes, it was my mistake, i create the extension with the context from-internal and everything went fine, now I'm having another problem, which is that I'm calling the echo-test extension, but asterisk hangs me 30 seconds later because sipp is not sending any RTP data. Any ideas on how to fix this. The demo context which is mentioned in [1] doesn't work. [1] http://www.rowetel.com/ucasterisk/ucasterisk.html Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniería de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://planeta.debianperu.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIPP problem
Hi everybody, I'm trying to load-test my Asterisk PBX using SIPP, but I always getting errors, I followed the instructions given in [1] which mainly was to create the user sipp in sip.conf and the dialing plan for his context in extensions.conf I'm using Asterisk 1.0.10 Any ideas or tutorial on how using SIP? [1] http://www.rowetel.com/ucasterisk/ucasterisk.html Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniería de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://planeta.debianperu.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Timeout Registration IAX2
Hi, I'm using IAX2 to connect remote users to my asterisk server. Both server and user are behind a nat. But sometimes the user registrates correctly but sometimes doesn't. Doing a debug i got: Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Acking anyway Sending 3 on 5/4132 to 200.31.126.250:4569 Sending 15 on 5/4132 to 200.31.126.250:4569 Received packet 0, (6, 13) Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Acking anyway Sending 3 on 5/4132 to 200.31.126.250:4569 Received packet 0, (6, 13) Packet arrived out of order (expecting 1, got 0) (frametype = 6, subclass = 13) Acking anyway Sending 3 on 2/4131 to 200.31.126.250:4569 Urgent handler Sending 10002 on 2/4131 to 200.31.126.250:4569 Where 200.31.126.250 is the public IP of the user. Any ideas what can be wrong? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniería de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://planeta.debianperu.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MySQL CDR
Hi everyone, I finished installing the Xorcom Rapid's Asterisk Packages with amportal (1.10.10), but i wasn't able to find the asterisk-mysql package. Any idea what happened there?, Is there another reposiitory for that package for asterisk 1.0.11. Or could somebody send me the cdr_addon_mysql.so file? Thanks for your responses, -- Diego Quintana a.k.a. RouterMaN Ingeniería de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://planeta.debianperu.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with tdm22b
Hi everyone, I have a problem installing interface card tdm22b in a debian etch machine. First I added manually the zaptel module: apt-get install zaptel-source kernel-headers-`uname -r` m-a a-i zaptel Then I do tumiwall:/usr/src# ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Slaves: 02) Channel 03: FXO Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) Does anybody knows why zaptel is not detecting the device?? My /etc/zaptel.conf contains: fxsks=1,4 fxoks=2,3 which I think is the correct order of my fxs and fxo Regards from Peru, -- Diego Quintana a.k.a. RouterMaN Estudiante Ing de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ http://routerman.blogsome.com http://planeta.debianperu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
